Packet-switched teleconferencing systems utilizing standard network communication protocols are becoming increasingly common. These systems may take advantage of the ubiquity and relatively low cost of network communication protocols to better offer teleconferencing services. However, as the popularity of teleconferencing services, and hence the number of users that may participate in a teleconference, has increased, the complexity of software and/or hardware necessary to mix together audio from many different sources to generate signals heard by the different users has increased accordingly. For example, in a system with multiple participants, one scheme may involve decoding and then mixing all the audio signals from all participants and subsequently transmitting the mixed audio signal back to the participants. One problem with this approach is that summing the audio signals from a large number of participants may cause saturation and/or distortion of the audio signal due to the digital representation, resulting in poor quality. A second problem with this approach is that audio signals from silent participants may contain only background noise, and adding signals from these participants can boost the level of background noise in the resulting mixed audio signal. Finally, a third problem with this approach is that decoding a large number of signals requires processing resources (e.g., as measured in millions of instructions per second (MIPS)) proportional to the number of users. There is thus a need to develop methods and/or systems to accommodate teleconferences involving larger groups of people by reducing complexity and/or improving speech quality.